<div dir="ltr">Hello,<div><br></div><div>I'm working on a one-way audio bug that may or may not be either Asterisk or Chrome WebRTC related. Details are here: <a href="https://code.google.com/p/webrtc/issues/detail?id=2347">https://code.google.com/p/webrtc/issues/detail?id=2347</a>, but basically the problem is that I get about 1 in 10 calls returning one way audio where audio flows from the PSTN to Chrome but not the other way.</div>
<div><br></div><div>I believe the issue may be with Asterisk 11, based on Wireshark traces and other troubleshooting, and if that's the case, I'm happy to try Freeswitch as an alternative.</div><div>I tried the demo here: <a href="https://webrtc.freeswitch.org/webrtc/portal.html">https://webrtc.freeswitch.org/webrtc/portal.html</a>, and the 20 test calls to the PSTN worked perfect with two way audio.</div>
<div><br></div><div>So, now, onto the Freeswitch question: Where is all the documentation for configuring the Websocket server on Freeswitch 1.4b? Where is the documentation that explains how to get ICE enabled? Why, when I enabled the ws-bind port in internal.xml does the client fail to establish a WS connection?</div>
<div><br></div><div>Basically, I'd love to get my hands on the docs for setting this up, but the Internet is largely silent on this matter? Is it possible for the Freeswitch team to share the configuration they used to get that JsSIP/Freeswitch demo setup? </div>
<div><br></div><div>If not the configuration, where would I find the docs for Freeswitch 1.4b?</div><div><br></div><div>Thank you!!<br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><i><b>"Every Call, Every Time!"</b></i></div>
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